similar to: Crosstalk - Is there a debug option for logging this?

Hi, Has anyone ever experienced crosstalk with sangoma analogue cards on 64bit? We have exhausted every test to try and replicate this and find a solution with Sangoma tech support, but we can not fix it. We are about to try the card and four *seperate* UK BT lines in a 32bit system. The current system is a 4gb, dual core cpu with pbx in a flash 1.4, Zaptel and Asterisk 1.4.21-2 Currently we

Hey List-ee's! We're on Asterisk 1.2 with a PRI/T1 and maybe 50 phones. Issue: When a conference is ongoing (meetme), if a call is placed to a PRI # (CALLER A) the audio from the conference is heard by that caller while the # they dialed is ringing in the background. The opposite also occurs where an outside caller will be heard in the conference channel as well. Frequency: 2-3 times

Hi, One other question I was wondering about. Is the reason that we hear the crosstalk with fec and packet loss percentage>0 is that Opus uses information from the left channel to try to error correct the right channel and vice versa? I am trying to understand the origin of the crosstalk. Thanks. -Jon > On Jan 27, 2017, at 12:29 PM, Jon Lederman <jon at soniccloud.com> wrote: >

On 27/01/17 12:29 PM, Jon Lederman wrote: > Thank you. Yes, we do need both channels independent. So, if we > encode each channel separately, we will be sacrificing the > compression ratio we would achieve with stereo encoding, correct? Not necessarily. Stereo makes two assumptions: 1) It assumes the two channels are somehow correlated 2) It assumes the two channels are meant to be

On 27/01/17 12:16 PM, Jon Lederman wrote: > When you say that with fec enabled, the threshold is increased, do > you mean the bitrate - i.e., you need higher bitrate with fec enabled > to suppress crosstalk? Correct. Another effect I forgot to mention is that Using FEC will actually force SILK/hybrid rather than CELT, so it's possible that disabling FEC makes you use CELT, which

Thanks. When you say that with fec enabled, the threshold is increased, do you mean the bitrate - i.e., you need higher bitrate with fec enabled to suppress crosstalk? Also, can you make any recommendations to decide whether to use fec or not? We are trying to tune the parameters appropriately. Audio quality and reduction of crosstalk are critical for us. Thanks again. Jon Sent from my

Hi, Every package I try to install in R i get the same error "had non zero-exit status". I use Ubuntu 18.04.1 LTS how can i fix this? Thank you This shows up when I try install.packages("Rcmdr", dep=T) Error : package ?htmltools? was installed by an R version with different internals; it needs to be reinstalled for use with this R version ERROR: lazy loading failed for

I've got a problem with a samba installation performed by another user so details of the installation are not known but hopefully someone out there has run into the same problem and can give me some information on a possible fix. One of our Samba servers seem to be doing a significant amount of crosstalk with a couple of our other Windows NT servers, I haven't been able to pinpoint what

I am making a battery of levelplots and wireframes for several fitted models. I wrote a function that takes the fitted model object as the sole argument and produces these plots. Various strange behavior ensued, but I have identified one very concrete issue (illustrated below): when my figure-drawing function includes the addition of points/lines to trellis plots, some of the

Hi Jean-Marc, Thank you. Yes, we do need both channels independent. So, if we encode each channel separately, we will be sacrificing the compression ratio we would achieve with stereo encoding, correct? So, based on what you say here is my understanding. Please confirm this is correct or not: 1) If we use fec, we can reduce cross-talk but increasing bitrate. However, that should result in

Thank you. Very helpful. > On Jan 27, 2017, at 12:40 PM, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > > On 27/01/17 12:29 PM, Jon Lederman wrote: >> Thank you. Yes, we do need both channels independent. So, if we >> encode each channel separately, we will be sacrificing the >> compression ratio we would achieve with stereo encoding, correct? > > Not

Ryan, from the console what does "zap show channel 1" or 2/3/4 in your case say. I have X100P's and I seem to be having similar sounding problems. I noticed that the above command shows the channel to be off-hook at all times when a phone line is plugged in. I don't know why or if it is a bug in the application reporting the status. dbc. Ryan Courtnage wrote: > On July 8,

Hi everyone, vmkit requires to know the size of a jitted method before emitting the method. This allows to allocate the correct size for the method. The attached patch creates this functionality when the flag SizedMemoryCode is on. In order to implement this functionality, i had to virtualize some MachineCodeEmitter functions. Is it OK to commit the patch? Thanks, Nicolas --------------

Hi, Is it possible to stream live, as opposed to recorded, OGG format yet? I had a look at the Icecast 2.0 beta instructions and it ends in a program called "example" which will stream a pre-recorded OGG file to the server. Is there anything to stream directly from the soundcard yet? If not, is there anything in the pipeline? If it is possible, what bitrates in mono/stereo are

Hi, I like Speex work. How can I contribute to echo cancelling ? Regards, Hubert

Occassionally on calls we get what sounds like low volume channel bleedover. Not clear enough to make out words, but not echo of either side of the main coversation. We're using a Digium card with 11 channels connected to PSTN lines. Any ideas on what the problem is or how to go about troubleshooting? Thanks! Paul

Hi Evan, Evan Cheng wrote: > 1) How are you computing size of the method being > jitted? I add a new pass with addSimpleCodeEmitter, with the emitter being a SizeEmitter. Since the target calls the emitter with functions such as writeByte, writeWord, etc.... the SizeEmitter class implements these function by incrementing a counter. At the end of the pass, the code size of the

> I looked at your network diagram. Try checking the configuration of the > Ethernet ports on the firewall and the Asterisk box. Make sure they are > set to auto-negotiate and not set to a fixed speed and fixed duplex. > I have found in the past that if one end of a link is expecting auto- > negotiation (as the switches probably are) and the other end is expecting > a fixed

Dear all, i try to plot with ggplot2. Therefor I have an matrix with 3 colums. With cbind I add an additional column called "col". I need this column "col" because in a later step and want to specify here some plot details which I will get from another analysis If I want to plot with this code, I have the problem that the legend is wrong. Blue changed to green and green to

This is almost certainly not an Asterisk-specific posting, but due to my inability to find a VoIP-focused Cisco list, I'll post here in the hopes of finding a more diverse user community. I am using a Cisco 7960 (version 6.0 SIP firmware) with Asterisk, and have been experiencing situations where the phone locks up. "Locks up" means that the bottom part of the screen